Acoustic dampening compensation system

ABSTRACT

At least one exemplary embodiment is directed to an acoustic management system configured to compensate for acoustical dampening comprising: a microphone configured to detect a first acoustic signal from an acoustic environment; and a logic circuit, where the logic circuit detects an onset of acoustical dampening between the acoustic environment and the microphone, and where the logic circuit generates an acoustic damping compensation filter, where the acoustic damping compensation filter is applied to the first acoustic signal generating a drive signal.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application claims the benefit of U.S. provisional patentapplication No. 60/893,617 filed on 7 Mar. 2007. The disclosure of whichis incorporated herein by reference in its entirety.

FIELD OF THE INVENTION

The present invention relates to acoustic signal manipulation, and moreparticularly, though not exclusively, to the acoustic compensation ofacoustic dampening by headwear on detected acoustic signals.

BACKGROUND OF THE INVENTION

Some acoustic detecting and/or measuring devices (e.g., earpieces, roommicrophones), that measure ambient acoustic signals can be adverselyaffected when an acoustic dampening occurs between the source of anacoustic signal in an environment and the detecting and/or measuringdevice. The effect can be frequency dependent and can adversely effectthe quality (e.g., spectral characteristics) of the measured acousticsignal.

SUMMARY OF THE INVENTION

At least one exemplary embodiment is directed to an acoustic managementsystem configured to compensate for acoustical dampening comprising: amicrophone configured to detect a first acoustic signal from an acousticenvironment; and a logic circuit, where the logic circuit detects anonset of acoustical dampening between the acoustic environment and themicrophone, and where the logic circuit generates an acoustic dampingcompensation filter, where the acoustic damping compensation filter isapplied to the first acoustic signal generating a drive signal.

At least one exemplary embodiment is directed to a method ofcompensating for acoustical dampening comprising: detecting a firstacoustic signal from an acoustic environment; detecting an onset ofacoustical dampening between the acoustic environment and a microphone;generating an acoustic damping compensation filter; and filtering thefirst acoustic signal using the acoustic damping compensation filter togenerate a drive signal.

At least one exemplary embodiment is directed to an acoustic dampeningdetection system comprising: a receiver to generate a first sound field;a microphone configured to measure an acoustic signal; and a logiccircuit, where if a magnitude of the measured acoustic signal averagedover the duration of the generation of the first sound field issubstantially equal to a reference magnitude, the logic circuitidentifies that an acoustic dampening event has occurred.

At least one exemplary embodiment is directed to a method of detectingacoustical dampening comprising: emitting a first acoustic wave;detecting a second acoustic wave over a time duration of the firstacoustic wave; comparing an averaged magnitude of the second acousticwave with a reference magnitude and determining if acoustical dampeningis present, where acoustical dampening is determined to be present ifthe averaged magnitude of the second acoustic wave is substantiallyequal to a reference magnitude.

At least one exemplary embodiment is directed to a method of detectingacoustical dampening comprising: emitting a first acoustic wave from anexternal receiver using a first electronic drive signal; detecting asecond acoustic wave over a time duration of the first acoustic wave;performing a cross-correlation between the first electronic drive signaland an electronic representation of the second acoustic wave; obtaininga measured magnitude of the cross-correlation at a lag-time; comparingthe measured magnitude with a reference magnitude value; and determiningif acoustical dampening is present, where acoustic dampening isdetermined to be present if the measured magnitude is greater than thereference magnitude value.

At least one exemplary embodiment is directed to an acoustic dampeningdetection system comprising: a microphone configured to measure anacoustic signal; and a logic circuit where if a rate of change of theacoustic signal at a time t is greater than a threshold value, theaverage value of the acoustic signal after time t has decreased anamount greater than a second threshold value, and the acoustic signalafter time t has an average value above a third threshold value thelogic circuit identifies that an acoustic dampening event has occurred.

Further areas of applicability of exemplary embodiments of the presentinvention will become apparent from the detailed description providedhereinafter. It should be understood that the detailed description andspecific examples, while indicating exemplary embodiments of theinvention, are intended for purposes of illustration only and are notintended to limited the scope of the invention.

BRIEF DESCRIPTION OF THE DRAWINGS

Exemplary embodiments of present invention will become more fullyunderstood from the detailed description and the accompanying drawings,wherein:

FIG. 1A illustrates one example of a acoustic dampening compensationdevice;

FIG. 1B illustrates one example of a situation of an acoustic dampeningelement affecting and acoustic signal;

FIG. 2A is a flow chart of an acoustic compensation system according toat least one exemplary embodiment;

FIG. 2B is a block diagram of a microphone signal conditioner;

FIG. 3A illustrates at least one method of detecting whether an acousticdampening event occurs in accordance with at least one exemplaryembodiment;

FIG. 3B illustrates at least one further method of detecting whether anacoustic dampening event occurs in accordance with at least oneexemplary embodiment;

FIG. 4A illustrates at least one further method of detecting whether anacoustic dampening event occurs in accordance with at least oneexemplary embodiment;

FIG. 4B illustrates a user voice spectral profile acquisition system inaccordance with at least one exemplary embodiment;

FIG. 5A illustrates a block diagram of a parameter look up system inaccordance with at least one exemplary embodiment;

FIG. 5B illustrates a headwear equalization system in accordance with atleast one exemplary embodiment; and

FIG. 6 illustrates an example of detecting a drop in sound pressurelevels using the rate of change, mean values, slopes and otherparameters in accordance with at least one exemplary embodiment.

DETAILED DESCRIPTION OF EXEMPLARY EMBODIMENTS OF THE PRESENT INVENTION

The following description of exemplary embodiment(s) is merelyillustrative in nature and is in no way intended to limit the invention,its application, or uses.

Exemplary embodiments are directed to or can be operatively used onvarious wired or wireless earpieces devices (e.g., earbuds, headphones,ear terminal, behind the ear devices or other acoustic devices as knownby one of ordinary skill, and equivalents).

Processes, techniques, apparatus, and materials as known by one ofordinary skill in the art may not be discussed in detail but areintended to be part of the enabling description where appropriate. Forexample specific computer code may not be listed for achieving each ofthe steps discussed, however one of ordinary skill would be able,without undo experimentation, to write such code given the enablingdisclosure herein. Such code is intended to fall within the scope of atleast one exemplary embodiment.

Additionally exemplary embodiments are not limited to earpieces, forexample some functionality can be implemented on other systems withspeakers and/or microphones for example computer systems, PDAs,Blackberrys, cell and mobile phones, and any other device that emits ormeasures acoustic energy. Additionally, exemplary embodiments can beused with digital and non-digital acoustic systems. Additionally variousreceivers and microphones can be used, for example MEMs transducers,diaphragm transducers, for examples Knowle's FG and EG seriestransducers.

Notice that similar reference numerals and letters refer to similaritems in the following figures, and thus once an item is defined in onefigure, it may not be discussed or further defined in the followingfigures.

At least one exemplary embodiment of the present invention isillustrated in FIG. 1A. The embodiment is a small headphone that isinserted in the ear of the user. The headphone consists of thesound-attenuating earplug 100 inserted into the ear. At the inner(eardrum-facing) surface of the plug, an ear-canal loudspeaker receiver102 is located for delivering the audio signal to the listener. At theouter (environment-facing) surface of the plug, an ambient-soundmicrophone 104 is located. Both the loudspeaker 102 and the microphone104 are connected to the electronic signal processing unit 106. Thesignal processing unit 106 also has a connector 108 for input of theaudio signal. Additionally, an ear-canal microphone 110 is placed at theinner (eardrum-facing) surface of the plug and an external loudspeaker112 is placed on the outer (environment-facing) surface of the plug forperforming other functions of the headphone system not described here(such as monitoring of sound exposure and ear health conditions,headphone equalization, headphone fit testing, noise reduction, andcustomization).

FIG. 1B illustrates an example of an acoustic dampening element 120 a,moving 140 a into the path of an acoustic signal or wave 130 a generatedby an acoustic source 100 a in ambient environment, The acoustic signalor wave 130 a can be acoustically damped to some level by 120 a, so thatthe acoustic signal measured by the microphone 110 a is effected.

FIG. 2 a depicts a general “top-level” overview of the Headwear acousticEqualization System (HEQS). Initialization of the HEQS 142 may bemanually invoked in a number of ways. One way is a manual activation; byeither the HEQS user (i.e. that person wearing the headset system inFIG. 1), or manually by a second person in a local or remote location(e.g. a supervisor). Another activation method is with an automaticmode, for instance in response to a loud sound or when the user donsheadwear (e.g. a helmet). There are a number of methods for detectingheadwear, as disclosed by the systems in FIGS. 3 a and 4 a. Whenheadwear detection systems determine that headwear is worn, thendecision unit 101 invokes a system 103 to determine the frequencydependant acoustic transmission index of the headwear (ATI_HW); themethod for to determine ATI_HW is described in FIGS. 5 a and 5 b. TheASM signal is then filtered 107 with a filter with a responseapproximating the inverse of ATI_HW 105. This gives a modified ASMsignal which approximates that ASM signal with the headwear removed. Thefilter system 107 may use entirely analog circuitry or may use digitalsignal processing, e.g. using an FIR-type digital filter. Depending onthe particular operating mode of the HEQS the ATI_HW may be updated on acontinuous or intermittent basis, as determined by decision unit 109. Ifthe operating mode is such that ATI_HW is calculated just once, then theupdate sequence is terminated 111.

FIG. 2 b describes an optional beam-forming platform. The beam formingplatform allows for the direction-dependant sensitivity of themicrophones in the headset in FIG. 1 to be electronically manipulated.For instance, the sensitivity may be increased in the direction of theHEQS users voice, and decreased in the direction of local noise sources,such as machine noise. The beam-forming platform 138 takes as its inputsat least three Ambient Sound Microphones (ASMs) 114, 122, 130. Theanalog signal is then amplified 116, 124, 132, and then filtered with aLow Pass Filter (LPF) 118, 126, 134 to prevent frequency aliasing by theAnalog to Digital Converters (ADC) 120, 128, 136. The beam-formingplatform 138 may also take as its input signal the output signal fromASMs in both the left and right headsets worn by the HEQS user. Theoutput signal 140 for each headset is considered the “conditioned ASMsignal” in other figures in the present invention.

FIG. 3 a depicts the SONAR-based headwear detection platform. Thissystem detects the presence of headwear using a SONAR-based system.Activation of this system 142 may be manually by a remote second person144 or by the HEQS user 141, or may be automatic 140 e.g. with acomputer timer. A SONAR test signal is reproduced with the ExternalReceiver (ER) 112 whilst simultaneously recording 143 the conditionedASM signal 148. The SONAR test signal 144 may be one of a number ofspecific test signals, as described in FIG. 3 b. The recorded ASM signalis analyzed 146 to extract the time-domain impulse response (IR) orfrequency domain transfer function 150. The frequency-domain transferfunction may be obtained empirically by dividing the spectral frequencyprofile of the SONAR test signal 144 by the spectral frequency profileof the recorded ASM signal 143 (if the spectral frequency profile islogarithmic, then this would be a subtraction of the two profiles).Alternatively, an adaptive filter such as one based on the LMS algorithmmay be used to iteratively approximate the time-domain impulse responseor frequency domain transfer function. If a maximum-length sequence(MLS) SONAR test signal is used, then the time-domain IR may be obtainedby cross-correlation of the MLS and recorded ASM signal 143. Theresulting IR is then analyzed to detect headwear. This is undertaken bydetecting features in the IR representative of strong sound reflectionsat time delays consistent with headwear; for instance, if a helmet isworn, then a reflection from the brim is expected at about 0.6 ms for abrim that is 10 cm from the headset. If close-fitting headwear is worn,such as a balaclava or fire-proof hood, then a higher-level IR would beobserved (especially at high frequencies) compared with the case when noheadwear is worn. If no headwear is worn, then decision unit 152determines that no additional filtering of the ASM signal is undertaken154. However, if the analysis of the obtained IR 146 predicts thatheadwear is worn, then depending on the particular operating mode 156(which may be set with the initialization system 142) filtering of theASM signal may be invoked with either a look-up table based EQ system(FIG. 5 a) or a voice-based EQ system (FIG. 5 b).

FIG. 3 b depicts the assembly for generating the SONAR test signal usedby the SONAR-based headwear detection platform in FIG. 3 b, and also forthe system which determines the acoustic transmission index of theheadwear described in FIG. 5 a. When the SONAR test signal is needed,the activation command 158 initializes a counter 160 which keeps arecord of the number of repetitions of the test stimulus (i.e. how manyaverages the analysis system makes). The particular test signal used maybe one of a number of signals; a frequency sweep 164 (ideally thisso-called chirp signal is from a lower frequency to a higher frequencywith a logarithmic rather than linear incremental sweep). Single ormulti-frequency sine-waves may also be used to give afrequency-dependant acoustic transfer function; a Maximum LengthSequence (MLS) signal 166 is often used to measure acoustic impulseresponses; transient (Dirac) impulses 168 give a IR directly; musicaudio 179 may be used to measure the transfer function, as well as noisebursts 171 which may be narrow-band filtered. Once the audio test signalis acquired 162, the signal is sent to the ER 178 via digital to analogconversion 174 and analog amplification 176 (which may befrequency-dependant to compensate for the electroacoustic sensitivity ofthe loudspeaker). A digital counter 180 tracks the number of times theaudio test signal is repeatedly reproduced with the ER, and decisionunit 182 terminates reproduction of the test signal 184 when the numberof repeats is sufficient.

Alternative to the SONAR-based system in FIG. 3 a is the Voice-basedheadwear detection platform described in FIG. 4 a. This system detectsthe presence of headwear using a user-generated voice. Activation ofthis system 142 may be manually by a remote second person 144 or by theHEQS user 141, or may be automatic 140 e.g. with a computer timer. Theheadwear is detected by analyzing the conditioned ASM signal 148 inresponse to user-generated voice 186. The prompting system for the userto speak is described in FIG. 4 b. The recorded ASM signal is analyzedby unit 143 when there is no headwear present to give a reference uservoice spectral profile 187. When the user dons headwear, they areprompted to speak (see FIG. 4 b) and a second ASM recording is made togive a current user voice spectral profile 188. The reference user voicespectral profile 187 and current user voice spectral profile 188 arecompared with unit 189 to give a transfer function which is analyzed topredict if headwear is worn. This analysis system may, for instance,determine that headwear is worn if the transfer function indicates thathigh-frequency content (e.g. at particular frequencies such as 1 kHz and4 kHz) are attenuated in the current user voice spectral profile 188compared with the reference user voice spectral profile 187 (e.g. are <5dB at these particular frequencies). If this analysis unit 189determines that headwear is not worn, then decision unit 152 does notfilter the ASM signal 154. Alternately, if analysis unit 189 determinesthat headwear IS worn, then decision unit 152 further determines thefrequency dependant acoustic transmission index of the headwear (ATI_HW)that is used to filter the ASM signal (i.e. with a filter responseapproximating the inverse of ATI_HW). ATI_HW is calculated depending onthe particular operating mode, as determined by unit 156, these twooperating modes are described in FIG. 5 a and FIG. 5 b.

FIG. 4 b describes the user-prompting system for the voice-basedheadwear detection platform. Either a pre-recorded verbal message 192 ornon-verbal message 194 (e.g. a tone) is reproduced with the ExternalReceiver 112 for the user to speak either a specific set of words (e.g.a phonetically balanced word list) or general words (e.g. normalconversation) or non-speech sounds (such as a whistle or hand-clap).This prompt may be repeated a number of times, according to theincremental repeat counter 196 and decision unit 198 which terminatesthe prompt message after a pre-defined number of repeated messageprompts.

FIG. 5 a describes a system for determining the acoustic transmissionindex of the headwear (ATI_HW). This is a frequency dependant value forthe free-field acoustic absorption of the headwear from an externalsound source to a measurement point on the other side of the headwear(specifically, measured at the entrance to the user's ear canal). Thesystem uses the SONAR headwear detection platform described in FIG. 3 ato obtain a headwear impulse response 150. It should be noted that thisis not the same as the ATI_HW; rather, it is the impulse responseobtained by emitting a SONAR test signal from the external receiver (112in FIG. 1 ) and recording the sound response at the ASM 104 (orconditioned ASM signal 140 in FIG. 2 b). In a particular optional learnmode 202, the IR of different headwear may be measured empirically, andtheir corresponding ATI_HW is also measured and stored in computermemory 204. The recently measured headwear IR 150 is then compared andmatch with measured IR's in the database 204 using matching unit 206(matching may be accomplished using a standard least mean squaresdifference approach). When the current headwear has been matched to onein the database, then the ASM signals 140 is filtered with an impulseresponse (or frequency-domain transfer function) which approximates theinverse of the matched ATI_HW 208. The filtering of the ASM signal byunit 210 may be accomplished using a digital FIR-type filter or anIIR-type digital filter, or a multi-band analog audio signal filter.Depending on the particular operating mode of the HEQS selected by theuser (or automatically selected) with selecting device 212, the ATI_HWmay be continually updated by decision unit 214.

FIG. 5 b describes an alternative method to that system in FIG. 5 a, fordetermining the ATI_HW of the headwear worn by the HEQS user. The methodin FIG. 5 b uses a measure of the user's reference voice spectralprofile 187; this is spectral profile of the (conditioned) ASM signalswhen no headwear is worn in response to user-generated speech ornon-speech (e.g. hand-claps). This is compared to the current ASMspectral profile when the user is wearing headwear 188. The comparisonis undertaken by unit 189, which may be a simple spectral subtraction(in the logarithmic, or deci-Bel, domain), or may be a division of thelinear spectral magnitude. The resulting transfer function approximatesATI_HW, and it's inverse is calculated by unit 220 to give a data vectorwhich can be used to filter the ASM signals with filter unit 210 (aspreviously described for FIG. 5 a).

FIG. 6 illustrates an acoustic signal 600 displayed in a non-limitingmanner as the sound pressure level versus time, t. In this non-limitingexample 600 is broken into three regions. The first region can becharacterized by an average value SPL-M1, with an associated baseline(e.g., a line fit utilizing least squares) having a slope SLP-1.Similarly the second and third regions can be characterized by anaverage value SPL-M2 and SPL-M3 respectively, with an associatedbaseline (e.g., a line fit utilizing least squares) having slopes SLP-2and SLP-3 respectively. FIG. 6 illustrates the situation where amicrophone (throughout the duration) is measuring the acoustic signal600, the measurement plotted in FIG. 6. At the onset of an acousticdampening event (e.g., sheet placed on microphone, headwear placed overearpiece microphone) the measured Sound Pressure Level (SPL) valuedecreases from SPL-M1 to SPL-M2 over a period of time Dt1. The rate ofdecrease, [(SPL-M2)-(SPL-M1)]/Dt1=R1, can be compared to a thresholdvalue T1 to aid in determining if an acoustic dampening event hasoccurred. For example if R1=20 dB/1 sec, and T1=10 dB/sec, and thecriteria for an acoustic dampening effect (e.g., rather than an acousticsource shut off) is |R1|<T1, then if |R1|<T1 (note that a criteria R1>T1can also be used as well as an equality relationship) as it is in theexample this can be used as an indication of an acoustic dampening eventrather than an acoustic source shut off. Note that in the exampleillustrated in FIG. 6, the acoustic dampening event is removed resultingin an increase from SPL-M2 to SPL-M3 in time Dt2. The rate of change,R2=[(SPL-M3)-(SPL-M2)]/Dt2, can be compared with a threshold T2 in asimilar manner as described above for T1. Another threshold that can beused is the dropped sound pressure levels (SPL-M2) average baselinevalue, for example if SPL-M2 >SPL-T3 then this can be used as anindication that an acoustic dampening event has occurred rather than anacoustic source shut off. For example if he threshold value SPL-T3 iseffective quiet (e.g., 80 dB) then if SPL-M2 drops to below SPL-T3 thenthis can be indicative of an acoustic source being turned off.

Other criteria can also be used as indicators of an acoustic dampeningevent occurring. For example if the slopes of the baselines before andafter shifting are significantly different this can be indicative of anacoustic source shut off rather than an acoustic dampening event. Forexample if |SLP-2-SLP-1|>|(SLP-1/2)| could be indicative that anacoustic source has been turned off and that possibly the slope of thesecond baseline (SLP-2) is close to zero.

FURTHER EXEMPLARY EMBODIMENTS

The following list various other exemplary embodiments of the invention.The list is meant as illustrative only not as a limitative list ofembodiments.

A self-contained Headwear Acoustic Equalization system (HEQS) tocompensate for the acoustic filtering of headwear (hats, helmets,fire-proof headwear etc) is herein described. The Headwear AcousticEqualization System (HEQS) empirically measures or determines theacoustic filtering properties of a head garment on a continuous,intermittent, or discrete basis. The acoustic filtering properties isused to compensate for the change in response of a microphone mounted onthe user's head (e.g. at or near the entrance to the ear canals) from anexternal sound source (e.g. voice) by filtering the microphone signalwith an audio signal filter (which may be adaptive or one from apre-defined filter database). The HEQS comprises:

-   -   A. An assembly to monitor the acoustic field in a Users        immediate environment using one or more Ambient Sound        Microphones (ASMs) located near to or at the entrance to one or        both occluded ear canals.    -   B. A signal processing circuit to amplify the signal from the        ASMs in (A) and to equalize for the frequency sensitivity of the        microphones and to low-pass filter (LPF) the signals prior to        digital conversion to prevent aliasing (with the cut-off        frequency of the LPF equal or less than half the sampling        frequency of the digital sampling system).    -   C. An analog-to-digital converter (ADC) to convert the filtered        analog signals in (B) to a digital representation.    -   D. An optional beam-forming platform that takes as its inputs        the digital signals from the ASMs from one or both headsets to        selectively affect the spatial sensitivity of the headset to        sound in the user's local environment.    -   E. An assembly to generate a desired SPL at or near the entrance        to one or both occluded (or partly occluded) ear canals        consisting of a loudspeaker receiver mounted in an earplug that        forms an acoustic seal of the ear canal. (This is the External        Receiver; ER).    -   F. A signal processing circuit to amplify the signal to the ER        to equalize for the frequency sensitivity of the transducer.    -   G. An digital-to-analog convert (DAC) to convert a digital audio        signal into an analog audio signal for reproduction with the ER.    -   H. A HEQS initialization system to start the HEQS; which may be        manually by the user with a voice-activation or with a physical        switch, or may be remote activation by a second person, or may        be automatically by a system which detects when headwear is        adjusted or fitted, or may be on a continuous or intermittent        basis.    -   I. A system to detect whether the HEQS user is wearing headwear.        Examples of headwear include: a military helmet, a SWAT hood,        balaclava, cold-weather face mask, helmet liner, neoprene        camouflage face mask, religious headwear such as a burka or        turban, or a fireproof face mask as typically worn by fighter        pilots and fire-service workers (fire men/ women).    -   J. A system to determine the frequency-dependant acoustic        attenuation of the headwear from an ambient sound source (such        as the user's voice or a sound-creating object in the        environment of the user) to the ASM(s). This attenuation        transmission index is called ATI_HW.    -   K. A system to filter the ASM signal with the inverse of the        ATI_HW of the headwear, so as to give an ASM signal similar to        that with the headwear absent.    -   L. A system to update the ATI_HW automatically on a continuous        basis.    -   M. A system to update the ATI_HW manually from either a        user-generated command or a command issued by a second remote        person.    -   N. A system to update the ATI_HW automatically on an        intermittent basis (e.g. every 10 minutes).    -   O. A system to transmit the ATI_HW to a data storage or analysis        system using a wired or wireless data transmission system.

Another embodiment of the invention enables the HEQS to automaticallydetermine if headwear is worn using a self-contained SONAR-basedheadwear detection platform. A SONAR test sound is emitted with anexternal receiver mounted on the headset device, and it's soundreflection is detected using one or more ambient sound microphonesmounted on the same headset. The reflected sound is analyzed todetermine the presence of headwear. This SONAR-based headwear detectionplatform comprises:

-   -   A. An assembly to monitor the acoustic field in a Users        immediate environment using one or more Ambient Sound        Microphones (ASMs) located near to or at the entrance to one or        both occluded ear canals.    -   B. A signal processing circuit to amplify the signal from the        ASMs in (A) and to equalize for the frequency sensitivity of the        microphones and to low-pass filter (LPF) the signals prior to        digital conversion to prevent aliasing (with the cut-off        frequency of the LPF equal or less than half the sampling        frequency of the digital sampling system).    -   C. An analog-to-digital converter (ADC) to convert the filtered        analog signals in (B) to a digital representation.    -   D. An optional beam-forming platform that takes as its inputs        the digital signals from the ASMs from one or both headsets to        selectively affect the spatial sensitivity of the headset to        sound in the user's local environment.    -   E. An assembly to generate a desired SPL at or near the entrance        to one or both occluded (or partly occluded) ear canals        consisting of a loudspeaker receiver mounted in an earplug that        forms an acoustic seal of the ear canal. (This is the External        Receiver; ER).    -   F. A signal processing circuit to amplify the signal to the ER        to equalize for the frequency sensitivity of the transducer.    -   G. An digital-to-analog convert (DAC) to convert a digital audio        signal into an analog audio signal for reproduction with the ER.    -   H. An initialization system to start the SONAR-based headwear        detection platform; which may be manually by the user with a        voice-activation or with a physical switch, or may be remote        activation by a second person, or may be automatically by a        system which detects when headwear is adjusted or fitted, or may        be on a continuous or intermittent basis.    -   I. A system to generate or retrieve from computer memory a SONAR        audio data test signal. This signal may be one of the following        types:        -   a. Swept sine “chirp” signal.        -   b. Maximum Length Sequence (MLS) test signal.        -   c. Dirac transient click signal.        -   d. Music audio signal.        -   e. Noise signal (white noise or link noise).    -   J. Circuitry to reproduce the audio test signal in (I) with the        external receiver.    -   K. A system to simultaneously record the ASM signal whilst the        test signal in (I) is reproduced with the ER.    -   L. A system to repeat the reproduction of the test signal in        (I).    -   M. A system to analyze the recorded ASM signal in response to        the SONAR test signal to determine if headwear is worn. This        system comprises a method to deconvolve the recorded ASM signal        to give a time domain impulse response or frequency domain        transfer function with reference to the original SONAR test        audio signal.    -   N. A system to determine if headwear is worn by analysis of the        deconvolved test impulse response (IR) or transfer function (TF)        in (M) with respect to a reference IR or TF made with no        headwear worn.

Another embodiment of the invention enables the HEQS to automaticallydetermine the frequency-dependant acoustic absorption characteristics ofthe headwear worn by a user (this is the Headwear acoustic AttenuationTransmission Index or ATI_HW). Once obtained, the ASM signal is filteredwith a filter corresponding to the inverse of ATI_HW. Thisself-contained SONAR-based headwear determination platform uses a SONARtest sound emitted with an external receiver mounted on the headsetdevice, and it's sound reflection is detected using one or more ambientsound microphones mounted on the same headset. The reflected sound isanalyzed to determine the headwear using a look-up table analysis withprevious measurements of known headwear. This SONAR-based headweardetermination platform comprises:

-   -   A. An assembly to monitor the acoustic field in a Users        immediate environment using one or more Ambient Sound        Microphones (ASMs) located near to or at the entrance to one or        both occluded ear canals.    -   B. A signal processing circuit to amplify the signal from the        ASMs in (A) and to equalize for the frequency sensitivity of the        microphones and to low-pass filter (LPF) the signals prior to        digital conversion to prevent aliasing (with the cut-off        frequency of the LPF equal or less than half the sampling        frequency of the digital sampling system).    -   C. An analog-to-digital converter (ADC) to convert the filtered        analog signals in (B) to a digital representation.    -   D. An optional beam-forming platform that takes as its inputs        the digital signals from the ASMs from one or both headsets to        selectively affect the spatial sensitivity of the headset to        sound in the user's local environment.    -   E. An assembly to generate a desired SPL at or near the entrance        to one or both occluded (or partly occluded) ear canals        consisting of a loudspeaker receiver mounted in an earplug that        forms an acoustic seal of the ear canal. (This is the External        Receiver; ER).    -   F. A signal processing circuit to amplify the signal to the ER        to equalize for the frequency sensitivity of the transducer.    -   G. An digital-to-analog convert (DAC) to convert a digital audio        signal into an analog audio signal for reproduction with the ER.    -   H. An initialization system to start the SONAR-based headwear        detection platform; which may be manually by the user with a        voice-activation or with a physical switch, or may be remote        activation by a second person, or may be automatically by a        system which detects when headwear is adjusted or fitted, or may        be on a continuous or intermittent basis.    -   I. A system to generate or retrieve from computer memory a SONAR        audio data test signal. This signal may be one of the following        types:        -   a. Swept sine “chirp” signal.        -   b. Maximum Length Sequence (MLS) test signal.        -   c. Dirac transient click signal.        -   d. Music audio signal.        -   e. Noise signal (white noise or link noise).    -   J. Circuitry to reproduce the audio test signal in (I) with the        external receiver.    -   K. A system to simultaneously record the ASM signal whilst the        test signal in (I) is reproduced with the ER.    -   L. A system to repeat the reproduction of the test signal in        (I).    -   M. A system to analyze the recorded ASM signal in response to        the SONAR test signal to determine if headwear is worn. This        system comprises a method to deconvolve the recorded ASM signal        to give a time domain impulse response or frequency domain        transfer function with reference to the original SONAR test        audio signal.    -   N. A system to determine if headwear is worn by analysis of the        deconvolved test impulse response (IR) or transfer function (TF)        in (M) with respect to a reference IR or TF made with no        headwear worn.    -   O. A system to determine what headwear is worn by the user by        comparing the empirically obtained IR or TR with a library of        measured IRs or TRs previously obtained. The empirically        obtained IR or TR is matched with the particular previously        measured IR or TR using, for example, the method of        least-squared difference.    -   P. A system to obtain the ATI_HW of the worn headwear using a        look-up table of previously measured ATI_HW's corresponding to        particular headwear IR's.    -   Q. A system to filter the ASM signal with a filter corresponding        to the inverse of the obtained ATI_HW. In an exemplary        embodiment, this filter is a digital FIR-type filter.

Another embodiment of the invention enables the HEQS to automaticallydetermine if headwear is worn using a self-contained Voice-basedheadwear detection platform. A Voice test sound is generated by the HEQSuser, and is simultaneously detected using one or more ambient soundmicrophones mounted on the same headset. In some embodiments theuser-generated sound is a non-voice sound such as a hand-clap or mouthwhistle. The measured sound is analyzed to determine the presence ofheadwear. This Voice-based headwear detection platform comprises:

-   -   A. An assembly to monitor the acoustic field in a Users        immediate environment using one or more Ambient Sound        Microphones (ASMs) located near to or at the entrance to one or        both occluded ear canals.    -   B. A signal processing circuit to amplify the signal from the        ASMs in (A) and to equalize for the frequency sensitivity of the        microphones and to low-pass filter (LPF) the signals prior to        digital conversion to prevent aliasing (with the cut-off        frequency of the LPF equal or less than half the sampling        frequency of the digital sampling system).    -   C. An analog-to-digital converter (ADC) to convert the filtered        analog signals in (B) to a digital representation.    -   D. An optional beam-forming platform that takes as its inputs        the digital signals from the ASMs from one or both headsets to        selectively affect the spatial sensitivity of the headset to        sound in the user's local environment.    -   E. An digital-to-analog convert (DAC) to convert a digital audio        signal into an analog audio signal for reproduction with the ER.    -   F. An initialization system to start the Voice-based headwear        detection platform; which may be manually by the user with a        voice-activation or with a physical switch, or may be remote        activation by a second person, or may be automatically by a        system which detects when headwear is adjusted or fitted, or may        be on a continuous or intermittent basis.    -   G. A system to obtain a Reference User Voice Profile (rUVP);        when activated by the system in (F), the rUVP acquisition system        works by the user generating some general or predefined verbal        messages (e.g. a collection of phonemically balanced words,        prompted by a messaging system reproduced with the ear canal        receiver). Alternatively, the user may be asked to generate        non-verbal sound stimuli, such as hand claps or mouth-whistles.        Whilst the user creates the Reference sound message, the ASM        signals are simultaneously recorded. The resulting spectral        profiles is the rUVP.    -   H. A system to obtain a Current User Voice Profile (cUVP); when        activated by the system in (F), the cUVP acquisition system        works by the user generating some general or predefined verbal        messages (e.g. a collection of phonemically balanced words,        prompted by a messaging system reproduced with the ear canal        receiver). Alternatively, the user may be asked to generate        non-verbal sound stimuli, such as hand claps or mouth-whistles.        Whilst the user creates the Reference sound message, the ASM        signals are simultaneously recorded. The resulting spectral        profiles is the cUVP.    -   I. A system to compare the rUVP and cUVP, and thus determine if        headwear is used. This comparison may be in the time domain, but        in an exemplary embodiment the comparison is in the frequency        domain. If the frequency content of the cUVP is less than the        rUVP at particular frequencies (e.g. ⅓^(rd) octave measurements        made at 1 kHz and 4 kHz) by a pre-defined amount (e.g. 5 dB),        then it may be deemed that headwear is currently being worn.

Another embodiment of the invention enables the HEQS to automaticallydetermine the frequency-dependant acoustic absorption characteristics ofthe headwear worn by a user (this is the Headwear acoustic AttenuationTransmission Index or ATI_HW). Once obtained, the ASM signal is filteredwith a filter corresponding to the inverse of ATI_HW. Thisself-contained Voice-based headwear determination platform uses a Voiceor non-voice (e.g. hand-clap) test sound created by the HEQS user, andis simultaneously recorded using one or more ambient sound microphonesmounted on a headset near to or in the user's ear canal. The recordedsound is analyzed to determine the particular headwear and it'scorresponding ATI_HW using a look-up table analysis with previousmeasurements of known headwear. This Voice-based headwear determinationplatform comprises:

-   -   A. An assembly to monitor the acoustic field in a Users        immediate environment using one or more Ambient Sound        Microphones (ASMs) located near to or at the entrance to one or        both occluded ear canals.    -   B. A signal processing circuit to amplify the signal from the        ASMs in (A) and to equalize for the frequency sensitivity of the        microphones and to low-pass filter (LPF) the signals prior to        digital conversion to prevent aliasing (with the cut-off        frequency of the LPF equal or less than half the sampling        frequency of the digital sampling system).    -   C. An analog-to-digital converter (ADC) to convert the filtered        analog signals in (B) to a digital representation.    -   D. An optional beam-forming platform that takes as its inputs        the digital signals from the ASMs from one or both headsets to        selectively affect the spatial sensitivity of the headset to        sound in the user's local environment.    -   E. An digital-to-analog convert (DAC) to convert a digital audio        signal into an analog audio signal for reproduction with the ER.    -   F. An initialization system to start the Voice-based headwear        detection platform; which may be manually by the user with a        voice-activation or with a physical switch, or may be remote        activation by a second person, or may be automatically by a        system which detects when headwear is adjusted or fitted, or may        be on a continuous or intermittent basis.    -   G. A system to obtain a Reference User Voice Profile (rUVP);        when activated by the system in (F), the rUVP acquisition system        works by the user generating some general or predefined verbal        messages (e.g. a collection of phonemically balanced words,        prompted by a messaging system reproduced with the ear canal        receiver). Alternatively, the user may be asked to generate        non-verbal sound stimuli, such as hand claps or mouth-whistles.        Whilst the user creates the Reference sound message, the ASM        signals are simultaneously recorded. The resulting spectral        profiles is the rUVP.    -   H. A system to obtain a Current User Voice Profile (cUVP); when        activated by the system in (F), the cUVP acquisition system        works by the user generating some general or predefined verbal        messages (e.g. a collection of phonemically balanced words,        prompted by a messaging system reproduced with the ear canal        receiver). Alternatively, the user may be asked to generate        non-verbal sound stimuli, such as hand claps or mouth-whistles.        Whilst the user creates the Reference sound message, the ASM        signals are simultaneously recorded. The resulting spectral        profiles is the cUVP.    -   I. A system to compare the rUVP and cUVP, and to determine the        particular headwear worn by the user. This comparison may be in        the time domain, but in an exemplary embodiment the comparison        is in the frequency domain. If the frequency content of the cUVP        is less than the rUVP at particular frequencies (e.g. ⅓^(rd)        octave measurements made at 1 kHz and 4 kHz) by a pre-defined        amount (e.g. 5 dB), then it may be deemed that headwear is        currently being worn. The transfer function of rUVP to cUVP is        compared to a database of measurements made with particular        headwear with a known Headwear acoustic Attenuation Transmission        Index or ATI_HW.

-   Alternative to the ATI_HW determination system in (I), a system to    empirically to determine ATI_HW which is calculated as the ratio of    rUVP to cUVP.    -   J. A system to filter the ASM signal with a filter corresponding        to the inverse of the obtained ATI_HW (i.e. obtained in process        I or J). In the at least one exemplary embodiment, this filter        is a digital FIR-type filter

While the present invention has been described with reference toexemplary embodiments, it is to be understood that the invention is notlimited to the disclosed exemplary embodiments. The scope of thefollowing claims is to be accorded the broadest interpretation so as toencompass all modifications, equivalent structures and functions of therelevant exemplary embodiments. Thus, the description of the inventionis merely exemplary in nature and, thus, variations that do not departfrom the gist of the invention are intended to be within the scope ofthe exemplary embodiments of the present invention. Such variations arenot to be regarded as a departure from the spirit and scope of thepresent invention.

1. An acoustic management system configured to compensate for acoustical dampening comprising: a microphone configured to detect a first acoustic signal from an acoustic environment; and a logic circuit, where the logic circuit detects an onset of acoustical dampening between the acoustic environment and the microphone, and where the logic circuit generates an acoustic damping compensation filter, where the acoustic damping compensation filter is applied to the first acoustic signal generating a drive signal.
 2. The system according to claim 1 further comprising: a receiver configured to emit a second acoustic signal, where the second acoustic signal is emitted in response to the drive signal.
 3. The system according to claim 2, where the microphone is an ambient sound microphone.
 4. The system according to claim 3, where the receiver is an ear receiver configured to direct the second acoustic signal to a user's ear.
 5. The system according to claim 4, where the acoustic damping compensation filter is frequency dependent.
 6. The system according to claim 5, where the acoustic damping compensation filter is added to a spectral signature of the first acoustic signal to generate a spectral signature of the drive signal.
 7. The system according to claim 3, where the ambient sound microphone is operatively connected to an earpiece.
 8. The system according to claim 7 where the earpiece is an earmuff.
 9. A method of compensating for acoustical dampening comprising: detecting a first acoustic signal from an acoustic environment; detecting an onset of acoustical dampening between the acoustic environment and a microphone; generating an acoustic damping compensation filter; and filtering the first acoustic signal using the acoustic damping compensation filter to generate a drive signal.
 10. The method according to claim 9, further comprising: sending the drive signal to a receiver, where the receiver generates a second acoustic signal.
 11. The method according to claim 10, where the acoustic damping compensation filter is frequency dependent.
 12. The method according to claim 11, where the step of filtering the first acoustic signal includes: generating a spectral signature of the acoustic damping compensation filter; and applying the spectral signature of the acoustic damping compensation filter to a spectral signature of the first acoustic signal.
 13. An acoustic dampening detection system comprising: a receiver to generate a first sound field; a microphone configured to measure an acoustic signal; and a logic circuit, where if a magnitude of the measured acoustic signal averaged over the duration of the generation of the first sound field is substantially equal to a reference magnitude, the logic circuit identifies that an acoustic dampening event has occurred.
 14. A method of detecting acoustical dampening comprising: emitting a first acoustic wave; detecting a second acoustic wave over a time duration of the first acoustic wave; comparing an averaged magnitude of the second acoustic wave with a reference magnitude and determining if acoustical dampening is present, where acoustical dampening is determined to be present if the averaged magnitude of the second acoustic wave is substantially equal to a reference magnitude.
 15. Method according to claim 14, further comprising: filtering the detected second acoustic wave using a band-pass filter. 16 A method of detecting acoustical dampening comprising: emitting a first acoustic wave from an external receiver using a first electronic drive signal; detecting a second acoustic wave over a time duration of the first acoustic wave; performing a cross-correlation between the first electronic drive signal and an electronic representation of the second acoustic wave; obtaining a measured magnitude of the cross-correlation at a lag-time; comparing the measured magnitude with a reference magnitude value; and determining if acoustical dampening is present, where acoustic dampening is determined to be present if the measured magnitude is greater than the reference magnitude value.
 17. Method according to claim 16, further comprising filtering the detected second acoustic wave using a band-pass filter.
 18. An acoustic dampening detection system comprising: a microphone configured to measure an acoustic signal; and a logic circuit where if a rate of change of the acoustic signal at a time t is less than a threshold value, the average value of the acoustic signal after time t has decreased an amount greater than a second threshold value, and the acoustic signal after time t has an average value above a third threshold value the logic circuit identifies that an acoustic dampening event has occurred. 